mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
amrnbdec: port to audiodecoder
This commit is contained in:
Mark Nauwelaerts
parent
2cb398d20d
commit
fa2d16b905
2 changed files with 93 additions and 238 deletions
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@ -96,19 +96,19 @@ static void gst_amrnbdec_set_property (GObject * object, guint prop_id,
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static void gst_amrnbdec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_amrnbdec_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_amrnbdec_setcaps (GstPad * pad, GstCaps * caps);
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static GstStateChangeReturn gst_amrnbdec_state_change (GstElement * element,
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GstStateChange transition);
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static void gst_amrnbdec_finalize (GObject * object);
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static gboolean gst_amrnbdec_start (GstAudioDecoder * dec);
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static gboolean gst_amrnbdec_stop (GstAudioDecoder * dec);
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static gboolean gst_amrnbdec_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static gboolean gst_amrnbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length);
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static GstFlowReturn gst_amrnbdec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_amrnbdec_debug, "amrnbdec", 0, "AMR-NB audio decoder");
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GST_BOILERPLATE_FULL (GstAmrnbDec, gst_amrnbdec, GstElement, GST_TYPE_ELEMENT,
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_do_init);
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GST_BOILERPLATE_FULL (GstAmrnbDec, gst_amrnbdec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER, _do_init);
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static void
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gst_amrnbdec_base_init (gpointer klass)
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@ -130,53 +130,53 @@ static void
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gst_amrnbdec_class_init (GstAmrnbDecClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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object_class->set_property = gst_amrnbdec_set_property;
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object_class->get_property = gst_amrnbdec_get_property;
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object_class->finalize = gst_amrnbdec_finalize;
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base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbdec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbdec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbdec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_amrnbdec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbdec_handle_frame);
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g_object_class_install_property (object_class, PROP_VARIANT,
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g_param_spec_enum ("variant", "Variant",
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"The decoder variant", GST_AMRNB_VARIANT_TYPE,
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VARIANT_DEFAULT,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrnbdec_state_change);
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}
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static void
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gst_amrnbdec_init (GstAmrnbDec * amrnbdec, GstAmrnbDecClass * klass)
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{
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/* create the sink pad */
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amrnbdec->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_setcaps_function (amrnbdec->sinkpad, gst_amrnbdec_setcaps);
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gst_pad_set_event_function (amrnbdec->sinkpad, gst_amrnbdec_event);
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gst_pad_set_chain_function (amrnbdec->sinkpad, gst_amrnbdec_chain);
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gst_element_add_pad (GST_ELEMENT (amrnbdec), amrnbdec->sinkpad);
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/* create the src pad */
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amrnbdec->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_use_fixed_caps (amrnbdec->srcpad);
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gst_element_add_pad (GST_ELEMENT (amrnbdec), amrnbdec->srcpad);
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amrnbdec->adapter = gst_adapter_new ();
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/* init rest */
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amrnbdec->handle = NULL;
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}
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static void
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gst_amrnbdec_finalize (GObject * object)
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static gboolean
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gst_amrnbdec_start (GstAudioDecoder * dec)
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{
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GstAmrnbDec *amrnbdec;
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GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
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amrnbdec = GST_AMRNBDEC (object);
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GST_DEBUG_OBJECT (dec, "start");
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if (!(amrnbdec->handle = Decoder_Interface_init ()))
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return FALSE;
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gst_adapter_clear (amrnbdec->adapter);
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g_object_unref (amrnbdec->adapter);
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amrnbdec->rate = 0;
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amrnbdec->channels = 0;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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return TRUE;
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}
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static gboolean
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gst_amrnbdec_stop (GstAudioDecoder * dec)
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{
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GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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Decoder_Interface_exit (amrnbdec->handle);
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return TRUE;
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}
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static void
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@ -214,13 +214,13 @@ gst_amrnbdec_get_property (GObject * object, guint prop_id,
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}
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static gboolean
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gst_amrnbdec_setcaps (GstPad * pad, GstCaps * caps)
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gst_amrnbdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
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{
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GstStructure *structure;
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GstAmrnbDec *amrnbdec;
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GstCaps *copy;
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amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
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amrnbdec = GST_AMRNBDEC (dec);
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structure = gst_caps_get_structure (caps, 0);
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@ -236,168 +236,86 @@ gst_amrnbdec_setcaps (GstPad * pad, GstCaps * caps)
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"rate", G_TYPE_INT, amrnbdec->rate, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
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amrnbdec->duration = gst_util_uint64_scale_int (GST_SECOND, 160,
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amrnbdec->rate * amrnbdec->channels);
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gst_pad_set_caps (amrnbdec->srcpad, copy);
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), copy);
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gst_caps_unref (copy);
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gst_object_unref (amrnbdec);
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return TRUE;
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}
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static gboolean
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gst_amrnbdec_event (GstPad * pad, GstEvent * event)
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static GstFlowReturn
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gst_amrnbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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GstAmrnbDec *amrnbdec;
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gboolean ret = TRUE;
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GstAmrnbDec *amrnbdec = GST_AMRNBDEC (dec);
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const guint8 *data;
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guint size;
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gboolean sync, eos;
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gint block, mode;
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amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
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size = gst_adapter_available (adapter);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_START:
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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gst_audio_decoder_get_parse_state (dec, &sync, &eos);
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/* need to peek data to get the size */
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if (gst_adapter_available (adapter) < 1)
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return GST_FLOW_ERROR;
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data = gst_adapter_peek (adapter, 1);
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/* get size */
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switch (amrnbdec->variant) {
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case GST_AMRNB_VARIANT_IF1:
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mode = (data[0] >> 3) & 0x0F;
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block = block_size_if1[mode] + 1;
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break;
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case GST_EVENT_FLUSH_STOP:
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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gst_adapter_clear (amrnbdec->adapter);
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amrnbdec->ts = -1;
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break;
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case GST_EVENT_EOS:
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gst_adapter_clear (amrnbdec->adapter);
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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break;
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case GST_EVENT_NEWSEGMENT:
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{
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GstFormat format;
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gdouble rate, arate;
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gint64 start, stop, time;
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gboolean update;
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gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
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&start, &stop, &time);
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/* we need time for now */
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if (format != GST_FORMAT_TIME)
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goto newseg_wrong_format;
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GST_DEBUG_OBJECT (amrnbdec,
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"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
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", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
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update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
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GST_TIME_ARGS (time));
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/* now configure the values */
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gst_segment_set_newsegment_full (&amrnbdec->segment, update,
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rate, arate, format, start, stop, time);
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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}
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case GST_AMRNB_VARIANT_IF2:
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mode = data[0] & 0x0F;
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block = block_size_if2[mode] + 1;
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break;
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default:
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ret = gst_pad_push_event (amrnbdec->srcpad, event);
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g_assert_not_reached ();
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return GST_FLOW_ERROR;
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break;
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}
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done:
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gst_object_unref (amrnbdec);
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return ret;
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GST_DEBUG_OBJECT (amrnbdec, "mode %d, block %d", mode, block);
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/* ERRORS */
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newseg_wrong_format:
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{
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GST_DEBUG_OBJECT (amrnbdec, "received non TIME newsegment");
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goto done;
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}
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*offset = 0;
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*length = block;
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_amrnbdec_chain (GstPad * pad, GstBuffer * buffer)
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gst_amrnbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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{
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GstAmrnbDec *amrnbdec;
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GstFlowReturn ret;
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guint8 *data;
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GstBuffer *out;
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amrnbdec = GST_AMRNBDEC (gst_pad_get_parent (pad));
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amrnbdec = GST_AMRNBDEC (dec);
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/* no fancy flushing */
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if (!buffer || !GST_BUFFER_SIZE (buffer))
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return GST_FLOW_OK;
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if (amrnbdec->rate == 0 || amrnbdec->channels == 0)
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goto not_negotiated;
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/* discontinuity, don't combine samples before and after the
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* DISCONT */
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if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
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gst_adapter_clear (amrnbdec->adapter);
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amrnbdec->ts = -1;
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amrnbdec->discont = TRUE;
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}
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/* the library seems to write into the source data, hence
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* the copy. */
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/* should not be a problem though */
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data = GST_BUFFER_DATA (buffer);
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/* take latest timestamp, FIXME timestamp is the one of the
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* first buffer in the adapter. */
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if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
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amrnbdec->ts = GST_BUFFER_TIMESTAMP (buffer);
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/* get output */
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out = gst_buffer_new_and_alloc (160 * 2);
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gst_adapter_push (amrnbdec->adapter, buffer);
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/* decode */
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Decoder_Interface_Decode (amrnbdec->handle, data,
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(short *) GST_BUFFER_DATA (out), 0);
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ret = GST_FLOW_OK;
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while (TRUE) {
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GstBuffer *out;
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guint8 *data;
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gint block, mode;
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/* need to peek data to get the size */
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if (gst_adapter_available (amrnbdec->adapter) < 1)
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break;
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data = (guint8 *) gst_adapter_peek (amrnbdec->adapter, 1);
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/* get size */
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switch (amrnbdec->variant) {
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case GST_AMRNB_VARIANT_IF1:
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mode = (data[0] >> 3) & 0x0F;
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block = block_size_if1[mode] + 1;
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break;
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case GST_AMRNB_VARIANT_IF2:
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mode = data[0] & 0x0F;
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block = block_size_if2[mode] + 1;
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break;
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default:
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goto invalid_variant;
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}
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GST_DEBUG_OBJECT (amrnbdec, "mode %d, block %d", mode, block);
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if (!block || gst_adapter_available (amrnbdec->adapter) < block)
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break;
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/* the library seems to write into the source data, hence
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* the copy. */
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data = gst_adapter_take (amrnbdec->adapter, block);
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/* get output */
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out = gst_buffer_new_and_alloc (160 * 2);
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GST_BUFFER_DURATION (out) = amrnbdec->duration;
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GST_BUFFER_TIMESTAMP (out) = amrnbdec->ts;
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if (amrnbdec->ts != -1)
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amrnbdec->ts += amrnbdec->duration;
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if (amrnbdec->discont) {
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GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
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amrnbdec->discont = FALSE;
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}
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gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbdec->srcpad));
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/* decode */
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Decoder_Interface_Decode (amrnbdec->handle, data,
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(short *) GST_BUFFER_DATA (out), 0);
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g_free (data);
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/* send out */
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ret = gst_pad_push (amrnbdec->srcpad, out);
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}
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gst_object_unref (amrnbdec);
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return ret;
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return gst_audio_decoder_finish_frame (dec, out, 1);
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/* ERRORS */
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not_negotiated:
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@ -407,57 +325,4 @@ not_negotiated:
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gst_object_unref (amrnbdec);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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invalid_variant:
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{
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GST_ELEMENT_ERROR (amrnbdec, STREAM, TYPE_NOT_FOUND, (NULL),
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("Invalid variant"));
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gst_object_unref (amrnbdec);
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return GST_FLOW_ERROR;
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}
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}
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static GstStateChangeReturn
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gst_amrnbdec_state_change (GstElement * element, GstStateChange transition)
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{
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GstAmrnbDec *amrnbdec;
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GstStateChangeReturn ret;
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amrnbdec = GST_AMRNBDEC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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if (!(amrnbdec->handle = Decoder_Interface_init ()))
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goto init_failed;
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (amrnbdec->adapter);
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amrnbdec->rate = 0;
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amrnbdec->channels = 0;
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amrnbdec->ts = -1;
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amrnbdec->discont = TRUE;
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gst_segment_init (&amrnbdec->segment, GST_FORMAT_TIME);
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:
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Decoder_Interface_exit (amrnbdec->handle);
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break;
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default:
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break;
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}
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return ret;
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/* ERRORS */
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init_failed:
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{
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GST_ELEMENT_ERROR (amrnbdec, LIBRARY, INIT, (NULL),
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("Failed to open AMR Decoder"));
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return GST_STATE_CHANGE_FAILURE;
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}
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}
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@ -21,7 +21,7 @@
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#define __GST_AMRNBDEC_H__
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/gstaudiodecoder.h>
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#include <interf_dec.h>
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G_BEGIN_DECLS
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@ -47,29 +47,19 @@ typedef enum
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} GstAmrnbVariant;
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struct _GstAmrnbDec {
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GstElement element;
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/* pads */
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GstPad *sinkpad, *srcpad;
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guint64 ts;
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GstAudioDecoder element;
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GstAmrnbVariant variant;
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GstAdapter *adapter;
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/* library handle */
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void *handle;
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/* output settings */
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gint channels, rate;
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gint duration;
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GstSegment segment;
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gboolean discont;
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};
|
||||
|
||||
struct _GstAmrnbDecClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioDecoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_amrnbdec_get_type (void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue